In digital radio broadcasts, signals are encoded in the digital domain, as opposed to traditional analog broadcasts using amplitude modulated (AM) or frequency modulated (FM) techniques. The received and decoded digital audio signals have a number of advantages over their analog counterparts, such as a better sound quality, and a better robustness to radio interferences, such as multi-path interference, co-channel noise, etc. Several digital radio broadcast systems that have been deployed and deployed, such as the Eureka 147 digital audio broadcasting (DAB) system and the in-band, on-channel (IBOC) DAB system.
Many radio stations that transmit digital radio also transmit the same radio programme in an analog manner, for example using traditional amplitude modulated (AM) or frequency modulated (FM) transmissions. When two broadcasts for the same radio programme are available (e.g., either two digital broadcasts, or one digital and one analog broadcast, of the same programme), there is the possibility that the radio receiver may switch or cross-fade from one broadcast to the other, particularly when the reception of one is worse than that of the other. Examples of such switching strategies, often referred to as ‘blending’, are described in U.S. Pat. No. 6,590,944 and US publ. No. 2007/0291876.
When a blending operation from one broadcast technique to another broadcast technique is performed, it is known that artefacts may appear during a cross-fade, if the signals are not perfectly aligned. For example, if there is a small delay between the signals, they will exhibit opposite phases at particular frequencies, and these frequencies will be cancelled out at some point during the cross-fade. This happens even if the delay is as small as two samples.
Furthermore, it is difficult to calculate delays between the signal samples accurately in such real-time systems, in order to determine and correct artefacts due to slightly mis-aligned broadcast signals, particularly if computational resources are restricted. In addition, computing of accurate sampling delay is especially difficult if the signals have different characteristics, e.g., because different pre-processing has been applied. During the cross-fade, there can also be signal cancellation due to phase inversion (i.e., the signals having opposite phase). Next to this, one of the signals may have undergone processing with non-linear phase (e.g., filtering with an infinite impulse response filter), which makes the delay between the signals frequency dependent, and makes it practically impossible to adapt the signals to be perfectly aligned.
When such blending operations occur, and when the FM signal is of sufficiently high quality but has switched to mono (say, because of its weak signal handling), there can be artefacts in the stereo image, especially when there are frequent transitions from the digital to the analog broadcast and back again. In addition to switching to mono, the weak signal handling may apply a high-cut filter to the FM signal, which can cause additional artefacts when switching between analog and digital broadcast.
When the reception quality of digital audio signal transmissions degrades, the received (encoded) signals may contain bit errors. If the bit errors are still present after all error detection and error correction methods have been applied, the corresponding audio frame may not be decodable anymore and is ‘corrupted’ (either completely or in part). One way of dealing with these errors is to mute the audio output for a certain period of time (e.g., during one or more frames). The left and right channel of a stereo transmission are encoded separately (or at least, for the most part), and a stereo signal is expected to remain a stereo one as the reception quality degrades.
When the reception quality of an FM tuner/signal deteriorates, the sum and difference signals are influenced differently. When the received FM signal contains white noise, the corresponding demodulated noise component linearly increases with frequency. Since the sum signal is present in the low frequency area (up to 15 kHz), the signal-to-noise ratio (SNR) is considerably better in the sum signal than in the difference signal (which is present in the band from 24 kHz to 53 kHz). This means that in noisy conditions, the sum signal contains less noise than the stereo signal (since the left and right signals are derived from the sum and the difference signal). Hence, when the reception quality of an FM transmission degrades, the audio signal is often changed from stereo to mono in order to preserve the audio quality of the sum signal. This operation exploits the fact that FM is transmitted as a sum and a difference signal, rather than as a left and a right channel.
From the above, it follows that two broadcasts, e.g., a DAB and an FM one, can have different stereo information, due to processing that has been performed as a result of bad reception quality. It can also be the case that the broadcasts have different stereo information under perfect reception conditions (e.g., AM has a lower audio bandwidth and is mono, so a hybrid DAB/AM combination will always have different characteristics). Therefore, when a blending operation from one broadcast to the other is performed, there can be stereo artefacts as a consequence, for example the stereo image will change during the blending operation, especially when there are frequent transitions from one broadcast to the other and back.
If the reception quality of the FM signal degrades further, a high-cut filter may be applied to the audio signal by the weak signal handling. The cut-off frequency of this filter is decreased with decreasing signal quality. The difference in high-frequency content between a digital and analog broadcast may also cause artefacts in blending, in particular with frequent transitions between the broadcasts.
These artefacts caused by weak signal handling (stereo and/or higher frequency information discarded on FM) can be reduced by using a long cross-fade time in the blending operation. This leads to a smoother, more gradual transition between the signals with different characteristics. In US20150371620 a mechanism is proposed that reduces the stereo artefacts by using different cross-fade times on sum and difference signals. Transitions in the sum signal can be done quickly, while the difference signals are cross-faded more slowly. This method with long cross-fade times requires that both broadcasts remain available for a sufficiently long time (preferably at least two seconds) after the start of the blending operation, in order to obtain a smooth cross-fading of the relevant signal characteristics. For DAB broadcasts this is not always possible: DAB signals can transition from good quality to being non-decodable from one frame to the next. If the DAB quality drops so abruptly, the slow cross-fade on the difference signal cannot be used, since the DAB signal is no longer available.
Thus, an improved audio processing circuit, audio unit and method of spectrum blending is needed.